Re: [Alsa-user] PXA270 + UCB1400 no sound
Kim wrote: $ cat sounds/alsa/Front_Center.wav /dev/dsp This device doesn't support .wav files; the sample rate will be wrong. Better try aplay. Anyway, you should get _some_ sound this way ... == ./card0/oss_mixer == VOLUME Master 0 ... if the volume is raised and unmuted. HTH Clemens - SF.Net email is sponsored by: Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Suzuki Hp-55 dijital piano with ALSA USB MIDI
S.Çağlar Onur wrote: I'm using ALSA 1.0.15 on 2.6.18. The MIDI device is detected and the needed modules hotplugs itself as soon as i plug the cable, but I'm not getting any data via the raw MIDI port or sequencer ports. Here is the output from dmesg when plugging the device: usb 2-1: new low speed USB device using uhci_hcd and address 7 The USB Audio specification does not allow low speed USB MIDI devices because MIDI transfer must use USB bulk transers, but bulk transfers do not exist in low speed mode. The 1.0.15 driver has a workaround to change the transfer type to interrupt transfers, ... ALSA usb/usbmidi.c:170: usb_submit_urb: -22 ... but it seems your kernel does not allow interrupt transfers on bulk endpoints. Try a different kernel. HTH Clemens - SF.Net email is sponsored by: Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] PXA270 + UCB1400 no sound
Thank you for your reply. This device doesn't support .wav files; the sample rate will be wrong. Better try aplay. Anyway, you should get _some_ sound this way ... Yes, Anyway, I want to get some sound! == ./card0/oss_mixer == VOLUME Master 0 ... if the volume is raised and unmuted. I checked the machine which could get a sound. It was the same setting ... Regards Kim Clemens Ladisch [EMAIL PROTECTED] wrote: Kim wrote: $ cat sounds/alsa/Front_Center.wav /dev/dsp This device doesn't support .wav files; the sample rate will be wrong. Better try aplay. Anyway, you should get _some_ sound this way ... == ./card0/oss_mixer == VOLUME Master 0 ... if the volume is raised and unmuted. HTH Clemens - SF.Net email is sponsored by: Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] M-Audio Audiophile USB and digital out
Hi Stephan Seitz! On 2007.12.16 at 17:08:25 +0100, Stephan Seitz wrote next: with too much apps, so better don't use it at all, and better yet, run jack or pulseaudio sound server (latter is more compatible one with general desktop setup, but you'd better be using very recent version of it) and don't use hw/plughw devices directly in other apps. I tried Pulseaudio server 0.9.5 (Debian Lenny) configuring it with Like I said, you shouldn't use such old version of pulseaudio. The first actually usable for general desktop setup is heavily patched 0.9.7, as it appeared in fedora 8; now 0.9.8 is available and includes all fixes in main tree. „load-module module-alsa-sink device=plughw:0,1 sink_name=output” and This is fine but not really required since pulseaudio should detect and use correct audio device by itself. Or at least you can switch it manually after startup with pavucontrol. If someone can tell me how to configure Pulseaudio I will try it. Install recent version and all the useful utilites and you'll be fine. Also it's a good idea to lock sampling frequency to 48k or 96k and increase resampling quality (resample-method = src-sinc-medium-quality), though some people may prefer no resampling at all (however things will become worse when mixing is at work then). -- Vladimir - SF.Net email is sponsored by: Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Distortion via SPDIF
Greetings list, I've just setup a new PC with an AD 1988B sound chip on the motherboard; it's hooked into my home theatre using an optical spdif cable. I can play audio files using DTS and AC3 passthrough direct to the receiver without issue. However, playing non encoded audio (MP3s, FLAC audio, etc) is distorted through all volume levels. It also doesn't make any difference if I have the IEC958 mixer level at 20% or 100%. I'm running ALSA 1.0.14 from Kernel 2.6.23, using the snd-hda-intel driver. It presents two devices on the card, I've set the PCM default to the digital device in asound.conf. # aplay -l List of PLAYBACK Hardware Devices card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: AD198x Digital [AD198x Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 What can I try to stop this distorted audio playback? Regards, Tom - SF.Net email is sponsored by: Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. http://ad.doubleclick.net/clk;164216239;13503038;w?http://sf.net/marketplace ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] File Plugin
I've spend so many days trying to make the file plugin to work. I've
read the pcm_plugins documentation and Everything everywhere, but no
luck..
here is my asound.conf
pcm.player {
type plug
slave {pcm fileplayer}
}
pcm.fileplayer {
type file
slave null
infile /home/carlos/record.wav
format raw
}
Can someone tell me how to correctly configure the file plugin to read
audio from a file?...
thanks
-
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